Hi, I have an avi file which contains audio/x-raw-int (and video, but my question is just about the audio). There is the caps: caps = audio/x-raw-int, endianness=(int)1234, channels=(int)2, width=(int)16, depth=(int)16, rate=(int)48000, signed=(boolean)true, codec_data=(buffer)1000000000000100000000001000800000aa00389b71 and (type: 118, taglist, audio-codec=(string)\"Uncompressed\\ 16-bit\\ PCM\\ audio\";) Using identity and -v, I can see that buffer duration is around 10 sec and the total is 20 sec. So there is only 2 audio buffers. Is there a gstreamer element that can change or split this buffer duration ? (Usually audio buffer duration is about 20 or 50 ms) There is also the "gst_query_set_latency" but what would be the inpact on the video (video buffer duration) ? Usually I configure the audio latency (=audio buffer duration) when using alsasrc, but how to do that with an avi file? Finally I can see : Implementation: Has getrangefunc(): gst_base_transform_getrange Has custom eventfunc(): gst_base_transform_src_event Has custom queryfunc(): 0xb7916800 Provides query types: (3): latency (Latency) in gst-inspect-0.10 audioresample So audioresample is able to only change the latency ? any example ? Sincerely Julien